Over two-thirds of the telecommunications facilities now in use for toll traffic digitally encode voice and voiceband data by sampling the customer signals at an 8000 Hz rate and encoding each sample into eight digital bits. This process of Pulse Code Modulation (PCM) provides an allocation of 64 kilobits per channel over 24 channels. Added framing information utilizes another 8 kilobits, thus resulting in the common transmission rate of 1.544 megabits per second. Various proposals have been made for increasing the available transmission channels; for example, by decreasing the number of encoding bits for each signal sample and thus substantially lessening the required transmission bandwidth.
The only one of these proposals that has reached significant implementation is the 32 kilobit Adaptive Differential Pulse Code Modulation (ADPCM) system which effectively provides a doubling of capacity to 48 channels at the usual sampling and transmission rates by encoding the customer signal samples into words of four bits each. In this system the values of the four encoding bits are assigned in PCM-to-ADPCM transcoders or in ADPCM digital channel banks by complex algorithms which both predict and adapt to the voiceband signals based upon their magnitude, waveform, frequency content, and frequency spread. The operation of an ADPCM encoding system may be seen in the general description in U.S. Pat. No. 4,513,426.
The relatively limited permutations in the signals derived from human speech make possible the successful operation of ADPCM in that this system can readily predict from previously occurring signal samples the probable magnitude and frequency range of the ensuing signal pattern, and can adapt to the usually moderate variations in these parameters by optimally selecting the encoding bit assignments which will narrowly encompass the pattern range. In this manner an ADPCM system can reasonably reproduce the waveforms associated with speech by means of only four encoding bits.
During telecommunication conversations, listeners are reasonable tolerant of noise occurring during speaking intervals if the channel is quiet between the speech bursts, and therefore will not find objectionable the four-bit quantizing noise that is generated in the ADPCM encoding process.
The implementation of voiceband data transmission, on the other hand, particularly with high-speed (9.6 kb/s) modem equipment, has created a problem in that these transmissions normally span all but a small fraction of the voice bandwidth and outstrip the capability of the ADPCM system to predict and adapt to their effectively random, wide bandwidth signals. The encoding system is thus unable to sense a trend in the signal to which the adaptive algorithm may be applied, and, therefore, generates simple four-bit encoding of the wide-band signal. The resulting high level of quantizing noise renders the telephone channel incapable of faithfully functioning as a medium for high-speed modem use.
Due to the serious degradation in transmission quality imposed by ADPCM encoding upon systems operating with high-speed modems or with other broad bandwidth applications, standards bodies have imposed restrictions, including complete exclusion, on the number of such systems that may be permitted on an end-to-end connection. Unfortunately, however, since they are based upon the use of test signals which have easily-predictable waveforms, none of the ANSI/IEEE Std. 743-1984 standard voiceband test procedures are capable of identifying the presence of an ADPCM system in a transmission circuit.
Thus, the capabilities of ADPCM encoding systems which enable them to predict and adapt to common voiceband conversation signals allow them to remain largely transparent to identification by previously available, standardized test methods. As a result, such standard test procedures cannot distinguish between a lower signal quality resulting from a single ADPCM system and that resulting from a number of less deleterious PCM systems in tandem, and are ineffective as a means for determining the quality of ADPCM equipment or the contribution such equipment may make to overall noise in a system.
The present invention, on the other hand, provides a method of testing which avoids the predictive and adaptive capabilities of an ADPCM encoding system in order to allow the identification of the presence of such a system in a voiceband telecommunications circuit, and which additionally enables accurate testing of the quality of ADPCM encoding equipment with respect to the generation of spurious noise in a transmission.